Pure sip meaning I would buy some DID's for the customers to call in and have it routed to my asterisk box via sip?
How reliable are those usually? On Nov 24, 2007 1:46 PM, Moshe Maeir <[EMAIL PROTECTED]> wrote: > Hi, > Why do you want to go the PRI route? > Why not pure SIP? That is the way we do it. Pretty easy to set up, no > initial investments in h/w and you > are not limited at all by the number of channels. > If you are interested, we can offer you your own partition on our server. > > Good Luck > Moshe Maeir > The Flat Planet Phone Co. > > > On Nov 24, 2007 7:41 AM, emist <[EMAIL PROTECTED]> wrote: > > > > > > > > Hey guys, > > > > I was hoping someone could clarify this for me since i've been trying to > > find out for a while now to no avail. Im thinking of deploying a call > > routing service through asterisk. Basically I want people to be able to > > call a number through the PSTN and then call whatever extension to be > > routed through a voip termination provider. > > > > Im guessing using a PRI is the best way to do this. However, im confused > > as to how it all works. Say a PRI has 23 usable channels, does that mean > > that I will be able to route 23 calls at the same time or does it mean > > that I would have to split 11 channels for incoming voice traffic(from > > PSTN) and 11 channels for outgoing voip traffic? > > > > Im stomped =\ > > > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-biz mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-biz > > > > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
